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Remove unused members from AudioDeviceBuffer

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Removes current_mic_level_, new_mic_level_ and clock_drift_, together
with APIs for accessing them.

Bug: webrtc:8598
Change-Id: I8e07396fcafd2a719e204730e2c7d26797bed762
Reviewed-on: https://webrtc-review.googlesource.com/39783
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21632}
This commit is contained in:
Fredrik Solenberg
2018-01-16 09:19:38 +01:00
committed by Commit Bot
parent d980c57c80
commit 1a50cd5894
10 changed files with 15 additions and 53 deletions
Download Patch File Download Diff File Expand all files Collapse all files

31
modules/audio_device/audio_device_buffer.cc
View File

@ -50,12 +50,9 @@ AudioDeviceBuffer::AudioDeviceBuffer()
play_channels_(0),
playing_(false),
recording_(false),
current_mic_level_(0),
new_mic_level_(0),
typing_status_(false),
play_delay_ms_(0),
rec_delay_ms_(0),
clock_drift_(0),
num_stat_reports_(0),
last_timer_task_time_(0),
rec_stat_count_(0),
@ -231,15 +228,6 @@ size_t AudioDeviceBuffer::PlayoutChannels() const {
return play_channels_;
}
int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level) {
#if !defined(WEBRTC_WIN)
// Windows uses a dedicated thread for volume APIs.
RTC_DCHECK_RUN_ON(&recording_thread_checker_);
#endif
current_mic_level_ = level;
return 0;
}
int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
RTC_DCHECK_RUN_ON(&recording_thread_checker_);
typing_status_ = typing_status;
@ -252,18 +240,11 @@ void AudioDeviceBuffer::NativeAudioInterrupted() {
recording_thread_checker_.DetachFromThread();
}
uint32_t AudioDeviceBuffer::NewMicLevel() const {
RTC_DCHECK_RUN_ON(&recording_thread_checker_);
return new_mic_level_;
}
void AudioDeviceBuffer::SetVQEData(int play_delay_ms,
int rec_delay_ms,
int clock_drift) {
int rec_delay_ms) {
RTC_DCHECK_RUN_ON(&recording_thread_checker_);
play_delay_ms_ = play_delay_ms;
rec_delay_ms_ = rec_delay_ms;
clock_drift_ = clock_drift;
}
int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
@ -307,15 +288,13 @@ int32_t AudioDeviceBuffer::DeliverRecordedData() {
}
const size_t frames = rec_buffer_.size() / rec_channels_;
const size_t bytes_per_frame = rec_channels_ * sizeof(int16_t);
uint32_t new_mic_level(0);
uint32_t new_mic_level_dummy = 0;
uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_;
int32_t res = audio_transport_cb_->RecordedDataIsAvailable(
rec_buffer_.data(), frames, bytes_per_frame, rec_channels_,
rec_sample_rate_, total_delay_ms, clock_drift_, current_mic_level_,
typing_status_, new_mic_level);
if (res != -1) {
new_mic_level_ = new_mic_level;
} else {
rec_sample_rate_, total_delay_ms, 0, 0, typing_status_,
new_mic_level_dummy);
if (res == -1) {
RTC_LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
}
return 0;